Introduction: Free Landline Using Google Voice and a RaspberryPi

Disclaimer: The following article is intended for users comfortable working on Linux based machines.

In this article I'll review the steps I used to configure a VoIP landline using a SIP interface through a Raspberry Pi based PBX with Freeswitch and Google Voice.  In other words, a free landline!

I spent the better part of the weekend reading wiki's, blogs, manuals and other posts and had to piece them all together to get my own personal solution working.  I thought it would be useful to write it up and share my experiences.

I've tried to capture my steps as best as I could .  Please let me know if anything does not work or needs to be amended.

NOTE: You do not have 911 capability with this solution.

Step 1: Things You'll Need


    - A Raspberry PI with "wheezy-raspbian" connected to your network.  Please see the Raspberry Pi Wiki for instructions on how to set this up.
    - A physical SIP device or software, like XLITE.  I'm using an old Sipura SPA-2100.  One end connected to your physical phone line and the other to your network.  If you decide to use a software solution, like XLITE, then this isn't a "true" landline replacement because you'll need your PC always on.  Using a SIP device, allows
    - A Google voice account.
   
     - Test your Google voice account, try calling a number from the web UI.
   

      Then log into Google Voice:
     -   Click on Settings > Phones
     -   Uncheck all phones
     -   Check Google Chat
     -   Log out of gmail ( Or turn off chat at the bottom of the gmail page)

Step 2: Get Started

Install dependencies on your Raspberry pi.

    #apt-get install autoconf automake gawk g++ git-core libjpeg62-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff4-dev libperl-dev libgdbm-dev libdb-dev libssl-dev

Download, Compile and install freeswitch. NOTE: This step takes a few hours to compile. 

    #mkdir /usr/local/freeswitch
    #useradd freeswitch -d /usr/local/freeswitch
    #chown -R freeswitch:freeswitch /usr/local/freeswitch

    #cd /usr/local/src
    #git clone git://git.freeswitch.org/freeswitch.git
    #cd /usr/local/src/freeswitch


    #./bootstrap.sh && ./configure --prefix=/usr/local/freeswitch && make clean && make clean modwipe && make && make install

    Make sure the following line is present and uncommented in /usr/src/freeswitch/modules.conf

    endpoints/mod_dingaling

    And build mod_dingaling:

    #make mod_dingaling-install

Make sure mod_dingaling is not commented out in file conf/autoload_configs/modules.conf.xml

     <load module="mod_dingaling"/>

Edit the conf/jingle_profiles/client.xml and replace all its contents with the following.  Then replace only the highlighted fields with your Gmail username and password.

    <include>
      <!-- Client Profile (Original mode) -->
      <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
    <include>
      <profile type="client">
        <param name="name" value="gtalk"/>
        <param name="login" value="YOUR_GMAIL@gmail.com/talk"/>
        <param name="password" value="GMAIL.PASSWORD"/>
        <param name="server" value="talk.google.com" />
        <param name="message" value="Thanks Google!" />
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        <param name="exten" value="2001"/>
        <param name="rtp-ip" value="auto"/>
        <param name="auto-login" value="true"/>
        <param name="sasl" value="plain"/>
        <param name="server" value="talk.google.com"/>
        <param name="tls" value="true"/>
        <param name="use-rtp-timer" value="false"/>
        <param name="vad" value="none"/>
        <param name="candidate-acl" value="wan.auto"/>
        <param name="local-network-acl" value="localnet.auto"/>
     </profile>

    </include>

Step 3: Start Freeswitch

Start freeswitch manually and test module

    #/usr/local/freeswitch/bin/freeswitch

You should get a console like in the picture .

Try running reload mod_dingaling  if you entered the correct credentials and all was compiled
correctly, you should see the following message:


+OK Reloading XML
+OK module unloaded
+OK module loaded

freeswitch@pbx> 2012-12-30 19:52:59.136376 [NOTICE] libdingaling.c:1674 XMPP server connected

2012-12-30 19:52:59.356369 [NOTICE] libdingaling.c:1686 XMPP authenticated


Exit the freeswitch console using the shutdown command, and return to the shell prompt.

Edit the conf/directory/default.xml and replace all its contents with the following.  Then replace  only the highlighted fields with your SIP device IP address

    <include>
      <!--the domain or ip (the right hand side of the @ in the addr-->
      <domain name="192.168.0.XXX">
        <params>
          <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
        </params>

        <variables>
          <variable name="record_stereo" value="true"/>
          <variable name="default_gateway" value="$${default_provider}"/>
          <variable name="default_areacode" value="$${default_areacode}"/>
          <variable name="transfer_fallback_extension" value="operator"/>
        </variables>

        <groups>
          <group name="public">
            <users>
              <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
            </users>
          </group>
        </groups>

      </domain>
    </include>


Create a file called conf/directory/default/2001.xml and paste the following contents, replacing only the highlighted fields with any random password (save it for later) and your name, which will be used for caller id.

    <include>
      <user id="2001">
        <params>
          <param name="password" value="MAKEUPONE"/>
          <param name="vm-password" value="1000"/>
        </params>
        <variables>
          <variable name="toll_allow" value="domestic,international,local"/>
          <variable name="user_context" value="default"/>
          <variable name="effective_caller_id_name" value="John Doe"/>
          <variable name="effective_caller_id_number" value="2001"/>
          <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
          <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
          <variable name="callgroup" value="default"/>
        </variables>
      </user>
    </include>

Step 4: Configure Your SIP Device

Login to your SIP (Or SIP software), and enter the following . (NOTE I will only demonstrate on the SPA-2100, other solutions will have different screens).   NOTE: your configuration page may vary based on your SIP solution.

    In the proxy (or domain) type in the IP address or hostname of your freeswitch PBX.
    The Display Name can be anything you want.
    User ID should be 2001 (because that is what I set it to).
    The passowrd  is from the one you made up above.



Start freeswitch on the command line /usr/local/freeswitch/bin/freeswitch
    Save and reboot the SIP device.  You SIP device should now be registered. You can test by way of hearing a dial tone on your phone.
    Open another login window to your PBX and leave the previous freeswitch console open in the the other window. We'll reference it later.
    Edit the conf/dialp/default.xml and add the following, right after the "unloop" extension section.  Then replace only the highlighted fields with your SIP device IP address.

         <extension name="gvoice_in">
          <condition field="source" expression="^mod_dingaling$">
            <!--<action application="info" />-->
            <action application="log" data="CONSOLE GV CALL IN!" />
            <action application="log" data="CONSOLE ${destination_number}"/>
            <action application="start_dtmf" />
            <action application="set" data="execute_on_answer=send_dtmf 1@2001"/>
            <!--<action application="cidlookup" data="$1"/>-->
            <action application="set" data="hangup_after_bridge=true" />
            <!--<action application="set" data="originate_continue_on_timeout=true"/>-->
            <!--<action application="set" data="call_timeout=35"/>-->
            <action application="bridge" data="user/2001@192.168.0.XXX"/>
            <action application="answer"/>
          </condition>
        </extension>

        <extension name="gvoice_out">
          <condition regex="any">
            <regex field="destination_number" expression="^(\d{10})$" />
            <regex field="dialed_extension" expression="^\+1(\d{10})@voice.google.com$" />
            <regex field="destination_number" expression="\+1(\d{10})$" />
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="set" data="ringback=${us-ring}"/>
            <action application="set" data="call_timeout=45" />
            <action application="ring_ready"/>
            <action application="bridge" data="dingaling/gtalk/+1$1@voice.google.com"/>
          </condition>
        </extension>

    Edit the conf/autoload_configs/dingaling.conf.xml and replace all its contents with the following.

        <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
          <settings>
            <param name="debug" value="0"/>
            <param name="codec-prefs" value="PCMU"/>
          </settings>

          <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>

        </configuration>


    Go back to the freeswitch console window and type reloadxml  then  reload mod_dingaling
    Now you are can test making outgoing and receiving calls.
    If all works to your satisfaction, its time to make freespace start automatically at boot time.

Creating a Freeswitch boot service

    Create a new file called /etc/init.d/freeswitch paste the contents from this file.

#chown -R freeswitch:freeswitch /etc/init.d/freeswitch

#chmod +x /etc/init.d/freeswitch

#update-rc.d  freeswitch defaults

#/etc/init.d/freeswitch start



    Reboot the pbx and check that the daemon started successfully and that everything is working.



Voice Codec Optimization (Optional)
I found that the default codec causes outgoing calls to be choppy.  To fix this, I changed the default codec to PCMA.

Replace the following in conf/vars.xml


<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>


with the following

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>


Save the file and stop and start the service

#service freeswitch stop  && sleep 15 && service freeswitch start


freeswitch startup script


#!/bin/sh
# Start/stop the freeswitch daemon.
#
### BEGIN INIT INFO
# Provides:          freeswitch
# Required-Start:    $network
# Required-Stop:
# Should-Start:
# Should-Stop:
# Default-Start:     2 3 4 5
# Default-Stop:  0 1 6
# Short-Description:
# Description:
### END INIT INFO

PATH=/bin:/usr/bin:/sbin:/usr/sbin
DESC="freeswitch"
NAME=freeswitch
DAEMON=/usr/local/freeswitch/bin/freeswitch
DARGS="-nc"
PIDFILE=/var/run/freeswitch.pid
SCRIPTNAME=/etc/init.d/"$NAME"

test -f $DAEMON || exit 0

. /lib/lsb/init-functions


# there because it should be in /etc/default/locale.
parse_environment ()
{
    for ENV_FILE in /etc/environment /etc/default/locale; do
        [ -r "$ENV_FILE" ] || continue
        [ -s "$ENV_FILE" ] || continue

         for var in LANG LANGUAGE LC_ALL LC_CTYPE; do
             value=`egrep "^${var}=" "$ENV_FILE" | tail -n1 | cut -d= -f2`
             [ -n "$value" ] && eval export $var=$value

             if [ -n "$value" ] && [ "$ENV_FILE" = /etc/environment ]; then
                 log_warning_msg "/etc/environment has been deprecated for locale information; use /etc/default/locale for $var=$value instead"
             fi
         done
     done



# Get the timezone set.
    if [ -z "$TZ" -a -e /etc/timezone ]; then
        TZ=`cat /etc/timezone`
    fi
}

# Parse the system's environment
if [ "$READ_ENV" = "yes" ] ; then
    parse_environment
fi


case "$1" in
start)  log_daemon_msg "Starting freeswitch " "freeswitch"
        ulimit -s 240
        OLDPID=`pgrep $NAME`
        NEWPID=`cat $PIDFILE` 2>/dev/null
        if [ "$OLDPID" =  "$NEWPID" ] ; then
                echo "Service already running " && exit 1
        else
                su - $NAME -c "$DAEMON $DARGS " >/dev/null 2>&1
        fi
        sleep 1
        ps -ef|grep $DAEMON|grep -v grep|awk '{print$2}' > $PIDFILE
        log_end_msg $?
        ;;
stop)   log_daemon_msg "Stopping freeswitch" "freeswitch"
        kill `cat $PIDFILE` >/dev/null 2>&1
        RETVAL=$?
        [ $RETVAL -eq 0 ] && [ -e "$PIDFILE" ] && rm -f $PIDFILE
        log_end_msg $RETVAL
        ;;
restart) log_daemon_msg "Restarting freeswitch" "freeswitch"
        $0 stop
        $0 start
        log_end_msg 0
        ;;
status)
        status_of_proc -p $PIDFILE $DAEMON $NAME && exit 0 || exit $?
        ;;
*)      log_action_msg "Usage: /etc/init.d/freeswitch {start|stop|status|restart|reload|force-reload}"
        exit 2
        ;;
esac
exit 0


Comments

author
SenorContento (author)2017-05-19

X-Lite no longer supports linux! It still supports Windows and Mac. However, the page at https://ubuntuforums.org/showthread.php?t=1492768 says you can use Ekiga on Linux!

author
Mumbles1 (author)2016-06-04

Obi 200 is a $39-$49 Google Voice to POT bridge that does this same thing. It takes about 10 min or less to setup up new google account and install OBi. Thats less than the cost of a pi and power supply and cable.

author
AdrianC152 (author)Mumbles12016-07-24

I am considering this because Obi charges $10/year subscription fee. They don't tell you this when you by the product, but after 1 year it stops working and they don't provide the information on how to correctly configure it unless you pay $10/year. So it would be nice to build your own and eliminate any third party between the device and Google Voice.

author
krisyow (author)AdrianC1522016-10-21

Obi doesn't technically charge a subscription fee nor for the firmware update, but does charge for the annual warranty (see here:

http://www.dslreports.com/forum/r30768637-READ-THIS-before-you-pay-10-to-Obihai-for-support).

It is however, a pain to go through switching to a LAN connection to get into the Obi's web client and update the firmware. The firmware is available for free (see here:

https://www.obitalk.com/forum/index.php?topic=9.0).

Paying the $10 for a year's worth of support and less than 5 minutes in firmware update time is less then the amount I spent trying to update it manually.

author
Mumbles1 (author)AdrianC1522016-07-24

I have 2 obi200's and an obi202 setup with google voice and obitalk thats been working flawlessly for almost 2 years now and Ive never paid for any subscription! One of the obi200's got hit by lightning about a year ago and the sent me an advanced replacement for free. I dont think that would of changed the "1year limit" for all 3 obi's

author
LisaJ90 (author)2016-08-11

These
are some great points and loads of information about my desired issue and I
liked 0800 call forwarding because
they have proved themselves really helpful by getting me the result I wanted!

author
amackenzie5 (author)2014-05-31

Any hints con how to hook up a speaker + mic to RPi to make VOIP calls, without needing a PBX on it? Skype would be great (but impossible?) Google Voice an option.

author

apt-get install linphone

man linphone

author
LaurenceC3 (author)2015-09-01

Rather than buying a Pi and also a SIP device you achieve the same using the following device http://www.amazon.com/OBi100-Telephone-Adapter-Service-Bridge/dp/B004LO098O

I have a couple of these and I just connect a standard cordless phone to the RJ11 and I have a free landline anywhere in the house. This is plug and play

author
shomas (author)2014-09-20

I ran into another issue

root@pi:/usr/local/src# git clone git://git.freeswitch.org/freeswitch.git
Cloning into 'freeswitch'...
fatal: unable to connect to git.freeswitch.org:
git.freeswitch.org[0: 198.22.64.251]: errno=Connection refused

A little research and I found this https://confluence.freeswitch.org/display/FREESWIT...
Stash not git: The git repository has been replaced by Stash. See the Makefile scripts for CentOS or Debian or the Developers page under Community for specifics.
A little more research and I found this
https://confluence.freeswitch.org/display/FREESWIT...
apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff5-dev libperl-dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev
cd /usr/src

# To build from Master, the latest source code:
git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
##### OR #####
# To build from the current release source code:
git clone -b v1.4 https://stash.freeswitch.org/scm/fs/freeswitch.git
cd /usr/src/freeswitch
# The -j argument spawns multiple threads to speed the build process
./bootstrap.sh -j
# if you want to add or remove modules from the build, edit modules.conf
vi modules.conf
# add a module by removing '#' comment character at the beginning of the line
# remove a module by adding the '#' comment character at the beginning of the line containing the name of the module to be skipped
./configure --enable-core-pgsql-support
make && make install

I notice that they didnt install libjpeg62-dev or libtiff4-dev. Mater of fact, they install libtiff5-dev along with a few others. I am speculating that the current release is higher than the one that you prepared in this instructable, along with the possibility that some modules left uncommitted wont need some of the extra packages. Bit I doubt that installing them anyways would be a problem.

I will follow the instructions as provided on the page for version 1.4 and merge them with your instructions "#./bootstrap.sh && ./configure --prefix=/usr/local/freeswitch && make clean && make clean modwipe && make && make install" as well as un-commenting
"endpoints/mod_dingaling" in /usr/local/src/freeswitch/modules.conf rather then /usr/src/freeswitch/modules.conf because the path you provide was not consistent with your prior instructions "cd /usr/local/src ; git clone git://git.freeswitch.org/freeswitch.git". I will respond back later with how well this work or not.

author
armantow123 (author)shomas2014-09-25

sir ca you help me on my raspberry pi pbx?

author
shomas (author)armantow1232014-09-25

I have been busy with life, and have not yet completed mine, but I will try as best I can. At what stage are you having a problem?

author
shomas (author)2014-09-20

Starting with a fresh system and after running
# apt-get update
# apt-get upgrade
and then running
# apt-get install autoconf automake gawk g++ git-core libjpeg62-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff4-dev libperl-dev libgdbm-dev libdb-dev libssl-dev

I get a bunch of "... is already the newest version"

and ended with
Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming.

The following information may help to resolve the situation:
The following packages have unmet dependencies: libtiff4-dev : Depends: libjpeg-dev

E: Unable to correct problems, you have held broken packages.

I proceeded to apt-get install each one at a time and then came to libtiff4-dev that apparently doesn't work with libjpeg62-dev

# apt-get install libtiff4-dev
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following extra packages will be installed: libjbig-dev libjpeg8-dev libtiffxx0c2
The following packages will be REMOVED: libjpeg62-dev
The following NEW packages will be installed: libjbig-dev libjpeg8-dev libtiff4-dev libtiffxx0c2
0 upgraded, 4 newly installed, 1 to remove and 0 not upgraded.
Need to get 633 kB of archives.
After this operation, 985 kB of additional disk space will be used.
Do you want to continue [Y/n]?

I selected Y and proceeded with the rest of the apt-get installs. I will get back to share if it works or not.

author
amackenzie5 (author)2014-05-31

Any hints con how to hook up a speaker + mic to RPi to make VOIP calls, without needing a PBX on it? Skype would be great (but impossible?) Google Voice an option.

author
Rainbowjeremy (author)2014-05-29

Will this work following the discontinued support for XMPP by google voice?

author
mcgann2113 (author)2014-05-25

Does this solution allow for incoming calls for free as well?

author
spiderstinger (author)2013-04-18

Which wheezy-raspbian image did you use to build this? What release date?

About This Instructable

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Bio: DIY home automation enthusiast.
More by hakk3r:Presence Light for Lync 2013/Skype for Business 2015Free Landline Using Google Voice and a RaspberryPi
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